1. Field of the Invention
The present invention relates to a signal processing apparatus and a signal processing method that allow editing a part of a digital signal that has been segmented as blocks each of which has a predetermined data amount and each block to be highly efficiently encoded along with an adjacent block.
2. Description of the Related Art
As a related art reference of a highly efficiently encoding method for an audio signal, for example, a transform encoding method is known. The transform encoding method is one example of a block-segmentation frequency band dividing method. In the transform encoding method, a time-base audio signal is segmented into blocks at intervals of a predetermined unit time period. The time-base signal of each block is converted into a frequency-base signal (namely, orthogonally transformed). Thus, the time-base signal is divided into a plurality of frequency bands. In each frequency band, blocks are encoded. As another related art reference, a sub band coding (SBC) method as an example of a non-block-segmentation frequency band dividing method is known. In the SBC method, a time-base audio signal is divided into a plurality of frequency bands and then encoded without segmenting the signal into blocks at intervals of a predetermined unit time period.
As another related art reference, a highly efficiently encoding method that is a combination of the band division encoding method and the SBC method is also known. In this highly efficiently encoding method, a signal of each sub band is orthogonally transformed into a frequency-base signal corresponding to the transform encoding method. The transformed signal is encoded in each sub band.
As an example of a band dividing filter used for the above-described sub band coding method, for example a QMF (Quadrature Mirror Filter) is known. The QMF is described in for example R. E. Crochiere “Digital coding of speech in sub bands” Bell Syst. Tech. J. Vol. 55. No. 8 (1976). An equal band width filter dividing method for a poly-phase quadrature filter and an apparatus thereof are described in ICASSP 83, BOSTON “Polyphase Quadrature filters—A new sub band coding technique”, Joseph H. Rothwiler.
As an example of the orthogonal transform method, an input audio signal is segmented into blocks at intervals of a predetermined unit time period (for each frame). Each block is transformed by for example a fast Fourier transforming (FFT) method, a discrete cosine transforming (DCT) method, or a modified DCT transforming (MDCT) method. As a result, a time-base signal is converted into a frequency-base signal. The MDCT is described in for example ICASSP 1987, “Sub band/Transform coding Using Filter Bank Designs Based on Time Domain Aliasing Cancellation”, J. P. Princen and A. B. Bradley, Univ. of Surrey Royal Melbourne Inst. of Tech.
On the other hand, an encoding method that uses a frequency division width in consideration of the hearing characteristics of humans for quantizing each sub band frequency component is known. In other words, so-called critical bands of which their band widths are proportional to their frequencies have been widely used. With the critical bands, an audio signal may be divided into a plurality of sub bands (for example, 25 sub bands). According to such a sub band coding method, when data of each sub band is encoded, a predetermined number of bits is allocated for each sub band. Alternatively, an adaptive number of bits is allocated for each sub band. For example, when MDCT coefficient data generated by the MDCT process is encoded with the above-described bit allocating method, an adaptive number of bits is allocated to the MDCT coefficient data of each block of each sub band. With the allocated bits, each block is encoded.
An example of a related art reference of such a bit allocating method and an apparatus corresponding thereto is described as “a method for allocating bits corresponding to the strength of a signal of each sub band” in IEEE Transactions of Acoustics, Speech, and Signal Processing, vol. ASSP-25, NO. 4, August (1977). As another related art reference, “a method for fixedly allocating bits corresponding to a signal to noise ratio for each sub band using a masking of the sense of hearing” is described in ICASP, 1980, “The critical band coder—digital encoding of the perceptual requirements of the auditory system”, M. A. Kransner MIT.
When each block is encoded for each sub band, each block is normalized and quantized for each sub band. Thus, each block is effectively encoded. This process is referred to as block floating process. When MDCT coefficient data generated by the MDCT process is encoded, the maximum value of the absolute values of the MDCT coefficients is obtained for each sub band. Corresponding to the maximum value, the MDCT coefficient data is normalized and then quantized. Thus, the MDCT coefficient data can be more effectively encoded. The normalizing process can be performed as follows. From a plurality of numbered values, a value used for the normalizing process is selected for each block using a predetermined calculating process. The number assigned to the selected value is used as normalization information. The plurality of values are numbered so that they increment by 2 dB of an audio level.
The above-described highly effectively encoded signal is decoded as follows. With reference to the bit allocation information, the normalization information, and so forth for each sub band, MDCT coefficient data is generated corresponding to a signal that has been highly efficiently encoded. Since a so-called inversely orthogonally transforming process is performed corresponding to the MDCT coefficient data, time-base data is generated. When the highly efficiently encoding process is performed, if the frequency band is divided into sub bands by a band dividing filter, the time-base data is combined using a sub band combining filter.
When normalization information is changed by an adding process, a subtracting process, or the like, a reproduction level adjusting function, a filtering function, and so forth can be accomplished for a time-base signal of which an encoded data has been decoded that is known as the editing method of data. According to this method, since the reproduction level can be adjusted by a calculating process such as an adding process or a subtracting process, the structure of the apparatus becomes simple. In addition, since a decoding process, an encoding process, and so forth are not excessively required, the reproduction level can be adjusted without a deterioration of the signal quality. In addition, in this method, an encoded signal can be modified without changing the time period of the generated signal by decoding, part of the signal generated by the decoding process can be changed with no influence from other parts.
In other than the method for changing normalization information, when the chronological relation between a decoded signal and an original signal (namely, a delay amount of phases) is obtained, encoded data that has the same chronological relation with a decoded signal can be generated.
When encoded data is changed in the above-described method, an editing operation such as a level adjustment can be performed corresponding to an increase or decrease of one value of normalization information (for example, 2 dB). Thus, such a level adjustment cannot be more precisely performed. In the chronological direction, an editing operation such as a level adjustment cannot be performed in the accuracy exceeding the minimum time unit corresponding to the encoding data format of the applied encoding method (the minimum time unit is for example, 1 frame).
Thus, due to such restrictions corresponding to the applied encoding method and encoding data format, the editing operations in the reproduction level and the frequency region and the editing operation in the chronological direction cannot be more accurately performed.